October 2007 - Posts - Andy's Blog
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Andy's Blog

October 2007 - Posts

  • The Last Yogurt

    Forgive me for my choice of title.  This has really nothing to do with yogurt, but more about the connections that people make in this crazy world of ours.

    My wife e-mailed me a tragic news story shortly after I arrived to work this morning.

    http://www.wthr.com/Global/story.asp?S=7244095

    For the past year my wife has been getting non-homogonized milk from an Amish farmer.  My wife has met him, and as he has had some health problems recently, she has been quite concerned.  I'm not much of a milk drinker, but the non-homogenized milk is easier on my kids stomachs, even my baby boy drinks it without any problems.  All of my kids prior to this had been fed soy or rice milk until they got old enough to where the cow's milk didn't bother them anymore (bright red butts and a diarrhea).  Not the best thing, since there is little or no fat in soy and rice milk.

    One day the delivery man who brings the veggies and milk from the Amish farmers out in Rockville, Indiana said he brought some plain yogurt for us to try (several local families meet once a week to pick up the delivery), and since I like yogurt very much, and since I haven't been happy with store-bought yogurt for many years now my wife brought it home.  Well we've been getting it ever since.  He milks the cows, he makes the yogurt.

    As you've probably already surmised, Melvin Fisher is the dairy farmer who was killed in yesterday's crash.

    I just sat down to lunch, and opened the lunch box my wife packed for me this morning.  And I just ate the last yogurt.

    If you would, please say a prayer for a little girl who was air lifted to Riley Children's Hospital, who will be missing her parents and three brothers very much.

     

     

     

  • AES 2007 – DSP-Based RIAA Filters (Attn: Vinyl Fans) - Part I

     

    Last weekend I got to attend the 123rd AES Convention in New York City.  The one paper presentation that perhaps would have the most interest among some of our resident KlipscHeads was one presented by R. S. Robinson of Channel D Corp. entitled “Filter Reconstruction and Program Material Characteristics Mitigating Word Length Loss in Digital Signal Processing – Based Compensation Curves used for Playback of Analog Signals.”  In layman’s terms that means “Don't worry about lost resolution when doing a RIAA filter using DSP.”  For those interested in reading it for yourself it is Convention Paper 7185, but unless you have access to the AES online library, that will be difficult.  After his presentation I asked him if this was available in some consumer product; it is… Virtual Vinyl has more information about it, and you can purchase the software to do this on your computer (if you have a Mac).

     

    The author also said to me (which I was rather surprised to hear) that many people use the DSP filter during playback of the actual vinyl LP… I was rather surprised at this since I would have assumed that most vinyl fans would rather play the LP once for archival purposes (saving wear on the irreplaceable recording), and then listen to the archived digital file through the DSP… or at the very least transfer it to CD…

     

    But I suppose that an extensive vinyl collection would take a lot of drive space… and the reason they are still listening to vinyl in the first place is that they believe vinyl is better than CDs.

     

    I will not reprint the article here, but I will attempt to give a synopsis.  I have not used this product and I am not endorsing it in any way.  Neither do I pretend to be a DSP Engineer.

     

    First a little something for those who don’t have a clue what I’m talking about…

     

    Before making an LP master, the treble is boosted (~20dB at 20kHz) and the bass is cut (~20dB @ 20Hz).  This is done to overcome limitations of the LP playback system (mainly the phono cartridge) and to pack more music onto a record (making al LP possible).  A phono preamp is then used to 1) amplify the weak signal from the phono cartridge (a.k.a. the stylus, or needle) and 2) to undo the RIAA equalization (boost the bass and cut the treble).

     

    Now back to the paper… The author’s intent is to apply DSP equalization for LP playback and archiving with the reasoning that the accuracy can exceed that of an analog design and avoids channel to channel variation.  He also believes restoration is facilitated since the click and pop removal can be done before equalization.

     

    A major consideration of any DSP processing is dynamic range and headroom. In the case of the LP you must consider a 20dB headroom allowance for the treble; likewise this implies the full digital dynamic range can’t be utilized for bass frequencies.  Taken together (40dB overall) this accounts for six to seven bits of potential loss of resolution.  The author claims that this analysis “omits the consideration of the reconstructive effects of the low frequency emphasis curve (digital filtering), plus the frequency balance of typical musical program material.”  Hence the title of his paper “Filter Reconstruction and Program Material Characteristics Mitigating Word Length Loss…”

     

    So what is word length loss?  If I have a digital word (for simplicity I’ll use 16 bits) say, 1101101100111011.  If I have the full dynamic range of a 16-bit ADC available I will have no losses, but if signal is attenuated (like in the RIAA curve) I can’t use all those bits.  If I attenuate the signal 40dB (multiply by 0.01) I will end up with (forgive me if I make any mistakes in my binary math) 0000001000110001.  If I simply apply scalar gain to this result I end up with 1101101100100100.  Notice the error in the last 5 bits.

     

    Perhaps it would be easier to see if I used a factor of 2…

     

    Start with

     

    1101101100111011 and divide by 128 you get…

    0000000110110110 (simply stated this is word length loss),

     

    and now multiply by 128, you get

     

    1101101100000000.  Compare that to the original…

    1101101100111011.  Notice the error in the last six bits.

     

    This is the same thing that would happen if you sampled the output of an LP (where 20Hz is 40dB down from 20kHz) and then applied scalar gain to the low frequencies, you’re losing resolution in the low frequencies.

     

    More to come next week…

    Posted Oct 12 2007, 02:49 PM by Andy W with 3 comment(s)
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  • Amplifier Classifications - Part II

     

    Class D Amplifiers

     

    The output devices are switched on and off at a very high frequency (compared to the desired output signal) and are driven with a PWM signal (Pulse Width Modulation).  This creates an output signal that is a square wave with a varying duty cycle.  The signal passes through a low-pass filter to remove the high frequency signal.  Since the output devices are fully on or fully off, the power dissipation is very low, and efficiency of the output stage can exceed 90% at full power in some cases.  Sometimes called a “Digital Amplifier” (usually a misnomer since a square wave is still an analog signal).

     

    Due to continuous advancement in semiconductor technology, class D amplifiers, once an oddity, are now thriving in the market.  Driven by the desire for smaller Hi-Fi (and not so Hi-Fi) systems and better efficiency, this segment will continue to grow for the foreseeable future.

     

    Many times class D amplifiers are called “digital amplifiers.”  I prefer to call them “switching amplifiers.”  There are some varieties that take a digital PCM signal and convert it directly to PWM, so technically it is not inaccurate, though it would in my opinion be technically inaccurate to call an analog input class D amplifier “digital.”  Nevertheless people still advertise “digital” amplifiers… because “digital” has been marketed to mean “better than analog” in everything from cell phones to cable to satellite TV to HDTV to HD radio.  “Better for whom?” is the question.

     

    Regardless, the current crop of IC class D amplifiers are very good performers, in terms of output power, efficiency, output noise, low distortion (rivaling good Class B amplifiers), and reliability.  There are also some high end designs that are even better.   With most class D amplifiers though, you have to watch out for changes in high frequency response.  This is due to the output filter, generally a second order filter.  The output filter is designed to be flat with a particular load impedance.  If a lower impedance is used the filter will be “overdamped” and there will be drop in high frequency output.  Conversely, if a higher impedance load is used the filter will be “underdamped” and there will be a rise in the high frequency response.  The response variation is generally not to severe, but if the amplifier is not loaded the output filter can “ring.”  Generally this is planned for, and doesn’t present a hazard to the amplifier.

     

    Class G Amplifiers

     

    This type of amplifier is basically a Class B amplifier with a second voltage rail.  When the signal is high enough the higher voltage rail is used, and when it is not needed, the lower voltage rail is used.  Due to the dynamic nature of music, this can reduce power dissipation over the Class B amplifier.

     

    Probably more of these amplifiers are used in Pro Audio than home audio, and the improvements in class D technology have pretty much rendered these amplifiers obsolete, in my opinion.  Class B (or AB if you must) might be next on the chopping block as Class D designs gain more acceptance.

     

    Class H Amplifiers

     

    This is similar to Class G except only one voltage rail is used, but the voltage of the rail is varied up and down with the input signal.  Generally this requires a switch mode power supply (otherwise there would be no point to using Class H, as a linear regulator would dissipate just as much power, if not more, than the Class B amplifier itself).

     

    BASH amplifiers as well as the Carver designs fit into this category.

     

    Bridged Amplifiers

     

    Any of the above amplifier types can be bridged (a BASH amplifier by design is always bridged).  To bridge a stereo amplifier the phase of the input to one of the channels is inverted and the speaker load is driven by both amplifier channels simultaneously.  The positive speaker lead is driven by the noninverted channel and the negative speaker lead is driven by the inverted amplifier channel.  The ground connection (or common connection) is not used.  There is more than one way to accomplish this, but the concept behind running a bridged amplifier is to create a “more” powerful amplifier.  I put “more” in quotes because the power supply and/or the output devices will ultimately determine how much power you can get.  If the power supply can’t deliver the required current, or if the output devices can’t handle the added stress, then the power output will be limited, and it’s possible that the output devices could be damaged.

     

    If under normal circumstances an amplifier is delivering 1Vrms to an 8 ohm loudspeaker, the speaker receives V2/R = 1/8 Watt of power.  If on the other hand the same amplifier is connected in a bridged arrangement, and the same 1Vrms is available at the noninverting output, there will be -1Vrms at the inverting output (or 1Vrms with an inverted phase, or 180° out of phase – pardon my nomenclature for the moment, it’s just to make the math easier).  The potential difference (voltage) at the speaker is now 1V - (-1V) = 2V.  2Vrms on an 8 Ohm speaker will now deliver V2/R = 4/8 = 1/2 Watt of power, or 4 times the power.  The voltage is doubled, and by Ohm’s law the current is doubled; Volt x Amps = Power and 2 x 2 = 4.  But notice again that the current is doubled, and unless the amplifier can deliver that current, that level of power will not be achieved.  The doubling of current is why many 4-8 Ohm rated amplifiers can only be used with 8 Ohm speakers (or higher) when operating in bridge mode; the doubling of current with a 4 Ohm load would exceed the design rating of the output devices, or the amplifier could overheat.

     

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